Audio Band Research Articles

Accurate smartphone indoor positioning using a WSN infrastructure and non-invasive audio for TDoA estimation

In this paper we propose a reliable acoustic indoor positioning system fully compatible with a conventional smartphone. The proposed system takes advantage of the smartphone audio I/O and its processing capabilities to perform acoustic ranging in the audio band using non-invasive audio signals and it has been developed having in mind applications that require high accuracy, such as augmented or virtual reality, gaming or audio guiding applications. The system works in a distributed operation mode, i.e. each smartphone is able to obtain its own position using only acoustic signals. In order to support the positioning system, a Wireless Sensor Network (WSN) of synchronized acoustic beacons is proposed. To keep the infrastructure in sync we developed an Automatic Time Synchronization and Syntonization (ATSS) protocol that ensures a sync offset error below 5 μs. Using an improved Time Difference of Arrival (TDoA) estimation approach (that takes advantage of the beacon signals’ periodicity) and by performing Non-Line-of-Sight (NLoS) mitigation, we were able to obtain very stable and accurate position estimates with an absolute error of less than 10 cm in 95% of the cases and a mean absolute standard deviation of 2.2 cm for a position refresh period of 350 ms.

Recognition of Music and Melody in Patients With Cochlear Implants, Using a New Programming Approach for Frequency Assignment

Introduction and objectiveMusic is a universal, cross-cultural phenomenon. Perception and enjoyment of music are still not solved with current technological objectives of cochlear implants. The objective of this article was to advance the development and validation of a method of programming of cochlear implants that implements a frequency allocation strategy. We compared standard programming vs frequency programming in every subject. MethodsWe studied a total of 40 patients with cochlear implants. Each patient was programmed with an optimal version of the standard program, using the Custom Sound Suite 3.2 cochlear platform. Speech tests in silence were performed using syllable word lists from the protocol for the assessment of hearing in the Spanish language. Patients implanted bilaterally were tested in both ears at the same time. For assessing music listening habits we used the Munich Music Questionnaire and “MACarena” (minimum auditory capability) software. ResultsAll patients achieved better results in recognition, instrument tests and tonal scales with frequency programming (P<.005). Likewise, there were better results with frequency programming in recognising harmonics and pitch test (P<.005). ConclusionsFrequency programming achieves better perception and recognition results in patients in comparison with standard programming. Bilateral stimulation patients have better perception of musical patterns and better performance in recognition of tonal scales, harmonics and musical instruments compared with patients with unilateral stimulation. Modification and frequency allocation during programming allows decreased levels of current intensity and increase the dynamic range, which allows mapping of each audio band less obtrusively and improves the quality of representation of the signal.

Low Distortion Switching Amplifier With Discrete-Time Click Modulation

An all-digital Class-D amplifier based on a discrete-time implementation of the click modulator is presented. The algorithm is able to generate binary signals with separated baseband, displacing the harmonic content produced by the modulation process above certain frequency chosen by the designer. Perfect demodulation can be achieved by a simple low-pass filter. Previous implementations of the discrete-time click modulator reported in the literature suffer from aliasing in the frequency domain. The approach proposed here avoids aliasing, without the necessity to increase (interpolate) the sampling frequency of the signals. Following a brief theoretical introduction, the performance of the proposed architecture is demonstrated by experimental measurements performed on an H-bridge amplifier. An 88 dB signal-to-noise ratio (SNR) and a total harmonic distortion (THD) + N less than 0.04% is attainable over the entire audio band, extending from 20 Hz up to 20 kHz; on the other hand, no traces of IMD appear above the predicted noise floor. These performance indices are obtained for switching rates as low as 40 kHz. The reduction of the switching frequency provides more flexibility for the design of the demodulation stage allowing to trade off between the complexity of the demodulation filter and the achievable efficiency of the switching stage.

Reconocimiento auditivo musical y melódico en pacientes con implante coclear, mediante nuevo método de programación de asignación frecuencial

Introduction and objectiveMusic is a universal, cross-cultural phenomenon. Perception and enjoyment of music are still not solved with current technological objectives of cochlear implants. The objective of this article was to advance the development and validation of a method of programming of cochlear implants that implements a frequency allocation strategy. We compared standard programming vs frequency programming in every subject. MethodsWe studied a total of 40 patients with cochlear implants. Each patient was programmed with a optimal version of the standard program, using the Custom Sound Suite 3.2 cochlear platform. Speech tests in quiet were performed using syllable word lists from the protocol for the assessment of hearing in the Spanish language. Patients implanted bilaterally were tested in both ears at the same time. For assessing music listening habits we used the Munich Music Questionnaire and «MACarena»(minimum auditory capability) software. ResultsAll patients achieved better results in recognition, instrument tests and tonal scales with frequency programming (P<.005). Likewise, there were better results with frequency programming in recognising harmonics and pitch test (P<.005). ConclusionsFrequency programming achieves better perception and recognition results in patients in comparison with standard programming. Bilateral stimulation patients have better perception of musical patterns and better performance in recognition of tonal scales, harmonics and musical instruments compared with patients with unilateral stimulation. Modification and frequency allocation during programming allows decreased levels of current intensity and increase the dynamic range, which allows mapping of each audio band less obtrusively and improves the quality of representation of the signal.

Low-frequency terrestrial gravitational-wave detectors

Direct detection of gravitational radiation in the audio band is being pursued with a network of kilometer-scale interferometers (LIGO, Virgo, KAGRA). Several space missions (LISA, DECIGO, BBO) have been proposed to search for sub-hertz radiation from massive astrophysical sources. Here we examine the potential sensitivity of three ground-based detector concepts aimed at radiation in the 0.1--10 Hz band. We describe the plethora of potential astrophysical sources in this band and make estimates for their event rates and thereby, the sensitivity requirements for these detectors. The scientific payoff from measuring astrophysical gravitational waves in this frequency band is great. Although we find no fundamental limits to the detector sensitivity in this band, the remaining technical limits will be extremely challenging to overcome.

A 0.8-V 230-$\mu$W 98-dB DR Inverter-Based $\Sigma \Delta$ Modulator for Audio Applications

This paper presents a ΣΔ modulator based on a gain-boost class-C inverter for audio applications. The gain-boost class-C inverter behaves as a low-voltage subthreshold amplifier and boosts its dc gain for the high-precision requirement. Meanwhile, an on-chip body bias is used to compensate the performance degradation of the inverter at a slow process corner or low supply voltage. The proposed inverter-based modulator is fabricated in a 65-nm mixed-signal CMOS process with a die area of 0.3 mm <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> . The experimental chip achieves 91-dB peak signal-to-noise-plus-distortion ratio (SNDR), 94-dB signal-to-noise ratio (SNR) and 98-dB dynamic range (DR) over a 20-KHz audio band with a 5-MHz sampling frequency and a 0.8-V supply voltage consuming only 230-μW power, which demonstrates that the gain-boost class-C inverter is particularly suitable for low-voltage micro-power high-resolution applications.

A reconfigurable upper audio band modem for data communication between mobile devices

For communicating short data sequences over small distances, the use of devices with conventional wireless radio frequency interfaces requires standardized hardware, dedicated infrastructure and appropriate Link/Network layer protocols. To address challenges associated with these requirements, a communication mechanism using devices which support simple audio interfaces (speakers and microphones) is proposed using the upper audio band (UAB) of frequencies (16---20 kHz). Devices with audio interfaces can be deployed in a personal area network for communicating at low data rates over small distances. Multi-tone FSK modulation is used for transmitting Reed---Solomon encoded data over the UAB. For peer-to-peer communication applications, a sensing mechanism is enabled on the receiving device to sense for empty time---frequency slots and schedule its data transmission at the appropriate times. A system prototype is developed using portable speakers and smartphones with sensitive microphones. The effective throughput of the modem is evaluated for different sensing durations and distances. Ad-hoc peer-to-peer networks can be enabled between mobile devices for communicating short data sequences based on the UAB modem.

A low-power inverter-based ΣΔ analog-to-digital converter for audio applications

This paper presents a design for a low-power, high-resolution audio ΣΔ analog-to-digital converter (ADC) based on a novel gain-boost class-C inverter. The gain-boost class-C inverter behaves as a sub-threshold amplifier, thereby minimizing power dissipation. The proposed ADC chip is fabricated in a SMIC 65-nm CMOS process with a die area of 0.63 mm2. With 1.2 V of supply voltage, the ADC chip achieves a peak signal-to-noise-plus-distortion-ratio (SNDR) of 92 dB and a dynamic range (DR) of 97 dB over the 20 kHz audio band, consuming only 1.13 mW. These results make the ADC particularly suitable for portable electronics applications.

QEEG Spectral and Coherence Assessment of Autistic Children in Three Different Experimental Conditions

We studied autistics by quantitative EEG spectral and coherence analysis during three experimental conditions: basal, watching a cartoon with audio (V–A), and with muted audio band (VwA). Significant reductions were found for the absolute power spectral density (PSD) in the central region for delta and theta, and in the posterior region for sigma and beta bands, lateralized to the right hemisphere. When comparing VwA versus the V–A in the midline regions, we found significant decrements of absolute PSD for delta, theta and alpha, and increments for the beta and gamma bands. In autistics, VwA versus V–A tended to show lower coherence values in the right hemisphere. An impairment of visual and auditory sensory integration in autistics might explain our results.

A Low-EMI 3-W Audio Class-D Amplifier Compatible With AM/FM Radio

A 3-W audio class-D amplifier compatible with AM/FM radio while requiring minimal external filtering is presented. The amplifier uses driver slew rate control to reduce electromagnetic interference (EMI) in the FM band and digital modulation to spread the PWM common-mode signal and place a null at the tuned AM channel. Two original feedback techniques are employed to linearize the H-bridge. One cancels signal-dependent propagation delays in the H-bridge to improve harmonic distortion and eliminate high-frequency noise folding into the audio band. The second technique boosts the power supply rejection ratio to simplify and reduce the cost of supply filters. The amplifier system, implemented in a standard 110-nm CMOS process, operates with a supply voltage ranging from 2.7 to 6.6 V and achieves an SNR of 95 dB and 85% efficiency.

A 65-nm low-power high-linearity ΣΔ ADC for audio applications

This paper presents a low-power high linearity ΣΔ analog-to-digital converter (ADC) for audio applications. By adopting low noise large output swing operational amplifiers in a 2-1 cascaded modulator, not only can the noise floor be reduced, but the input signal range can also be enlarged. A low-power, area-efficient digital decimation filter was also designed to decrease the area and the power cost. The ADC was fabricated in the SMIC 65 nm single-poly-eight-metal (1P8M) mixed-signal complementary metal-oxide-semiconductor (CMOS) process with a die area of 0.36 mm2. Measurement results showed that a 90 dB peak signal to noise plus distortion ratio (SNDR) and a 93 dB dynamic range (DR) were achieved over the 22.05 kHz audio band. The power dissipation was 2.2 mW from 1.2 V power supply, which is suitable for audio codec applications.

A 15-bit 140-&lt;formula formulatype="inline"&gt;&lt;tex Notation="TeX"&gt;$\mu$&lt;/tex&gt; &lt;/formula&gt;W Scalable-Bandwidth Inverter-Based &lt;formula formulatype="inline"&gt; &lt;tex Notation="TeX"&gt;$\Delta \Sigma $&lt;/tex&gt;&lt;/formula&gt; Modulator for a MEMS Microphone With Digital Output

A discrete-time audio ΔΣ modulator for a MEMS microphone with digital output is presented that features a scalable signal bandwidth to also support ultrasonic frequencies for proximity sensing applications such as gesture recognition. The modulator achieves low power consumption and a small die area by using simple digital inverter instead of OTAs. To increase the robustness of the inverter approach, an LDO is used to provide an internally generated supply which regulates the inverter bias point. The LDO also improves the power supply rejection of the pseudo-differential architecture to above 78 dB. The presented ΔΣ modulator supports a scalable signal bandwidth up to 100 kHz, by dynamically adjusting the internal supply voltage depending on the sampling frequency. In the nominal 20-kHz audio band, it dissipates 140 μW from a minimum 1.5-V supply and achieves a DR of 92.6 dB , a SNDR of 87.9 dB, and a THD of -102.7 dB, respectively.

Gravitational wave astronomy

We are entering a new era of gravitational-wave astronomy. The ground-based interferometers have reached their initial design sensitivity in the audio band. Several upper limits have been set for anticipated astrophysical sources from the science data. The advanced detectors in the US and in Europe are expected to be operational around 2015. New advanced detectors are also planned in Japan and in India. The first direct detections of gravitational waves are expected within this decade. In the meanwhile, three pulsar timing array projects are forming an international collaboration to detect gravitational waves directly in the nanoHertz range using timing data from millisecond pulsars. The first direct detection of nanoHertz gravitational waves are also expected within this decade. In this paper, we review the status of current gravitational-wave detectors, possible types of sources, observational upper limits achieved, and future prospects for direct detection of gravitational waves.

Filterless Class-D Amplifier With Pseudorandomized Carrier Frequency Modulation for EMI Reduction

This paper presents a filterless Class-D amplifier with pseudorandomized carrier frequency modulation (PRCFM) to mitigate electromagnetic interference (EMI) issue associated with the removal of an off-chip LC filter. The designed filterless Class-D amplifier occupies a total area of 1 mm2 and is fabricated in 0.18-μm CMOS process technology. The proposed PRCFM circuit occupies only less than 1% of the IC area. Measurement results have demonstrated the effectiveness of the PRCFM scheme for EMI reduction. The PRCFM scheme consumes very little additional power and therefore does not affect the power efficiency of the amplifier. In addition, the proposed scheme also maintains very good total harmonic distortion over the audio band of interest (100 Hz to 6 kHz) and modulation index range.

IN-EAR DIGITAL ELECTRONIC NOISE CANCELLING AND COMMUNICATION DEVICE

A noise canceling and communication system is described. An in-ear device is adapted to fit in the ear canal of a device user. A passive noise reduction element reduces external noise entering the ear canal. An external microphone senses an external acoustic signal outside the ear canal to produce a representative external microphone signal. An internal microphone senses an internal acoustic signal proximal to the tympanic membrane to produce a representative internal microphone signal. One or more internal sound generators produce a noise cancellation signal and an acoustic communication signal, both directed towards the tympanic membrane. A probe tube shapes an acoustic response between the internal sound generator and the internal microphone to be relatively constant over a wide audio frequency band. An electronics module is located externally of the ear canal and in communication with the in-ear device for processing the microphone signals using a hybrid feed forward and feedback active noise reduction algorithm to produce the noise cancellation signal. The noise reduction algorithm includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the noise cancellation signal for fit and geometry of the ear canal of the user. The communication component also includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the communication signal for fit and geometry of the ear canal of the user and to assure that the communication signal does not interfere with the noise reduction algorithm and that the noise cancellation signal does not interfere with passing of the communication signal.

Thin-Film Piezoelectric MEMS Transducer Suitable for Middle-Ear Audio Prostheses

We report a thin-film piezoelectric microelectromechanical systems (MEMS) transducer suitable for in vivo implantation as audio prosthesis and a method for fabricating it. The transducer consists of a layered stack of thin films comprising a MEMS membrane made of an isolating oxide interface; two platinum (Pt) layers acting as electrodes, electrical paths, and contacting pads; a thin-film sputtered aluminum nitride (AlN) layer acting as acoustic active material of the transducer; a passivation layer for device protection; and a 3-D micromachined acoustic cavity fabricated on the bulk of a silicon-on-insulator supporting substrate. The transducer has a reduced size and a low cost associated with CMOS-like mass production. Finite-element modeling and experimental tests conducted at the audio and ultrasonic bands show that the device has the performance required in audio prosthesis applications. The technology offers possibilities for biocompatibility or biocompatible packaging and is integrable to microelectronic circuit technologies. <formula formulatype="inline" xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex Notation="TeX">$\hfill$</tex></formula> [2012-0004]

An 8 $\Omega$ 2.5 W 1%-THD 104 dB(A)-Dynamic-Range Class-D Audio Amplifier With Ultra-Low EMI System and Current Sensing for Speaker Protection

Class-D amplifiers for portable devices are now demanding more than 2 W 1% THD output power on 8 Ω speakers in smartphone and tablet for a better multi-media user experience. However higher acoustic power brings a lot of additional constraints such as speaker damage, more EMI energy and battery reset. The presented paper demonstrates a 2.5 W 1% THD into 8 Ω Class-D amplifier with 104 dB(A) dynamic range using a new power stage architecture to reduce silicon area. The design includes a ULEMI system to reduce high frequency energy without impact on audio band noise and an embedded load current sensing for speaker protection algorithm.

오디오 D/A 컨버터를 위한 인터폴레이티드 디지털 델타-시그마 변조기

디지털 입력 D급 증폭기는 보청기에서 사용되고 있으며 D급 증폭기는 디지털 회로와 아날로그 회로로 구성되어진다. 아날로그 회로는 가청 주파수 대역에서 잡음을 억제하고 디지털 입력을 아날로그 신호로 변환한다. 본 논문에서 제안한 인터폴레이티드 디지털 델타-시그마 변조기는 디지털 신호 처리기의 출력 신호를 D/A 변조기 입력에 적합하도록 데이터를 변조시킨다. 디지털 필터는 16-bit, 25-kbps 펄스 코드 변조 신호를 16-bit, 50-kbps 신호로 보간 작업을 한다. 이 보간 필터 출력은 3차 디지털 델타-시그마 변조기를 통하여 노이즈 쉐이핑(noise shaping) 처리된다. 최종적으로, 1.5-bit, 3.2-Mbps 신호가 D/A변조기 입력으로 인가된다.

Third-order parametric array generated by distantly spaced primary ultrasonic tones

Traditional parametric arrays are produced by a second-order nonlinear interaction between two primary ultrasonic tones that are close in frequency, resulting in a difference tone that is in the audio band. This article presents a parametric array produced by a third-order nonlinear interaction between two primary ultrasonic tones that are distantly spaced in frequency such that one tone is approximately the second harmonic of the other. The result is a third-order lower intermodulation (IM3) tone in the audio band with greater directivity and lower side lobe amplitude than comparable second-order fields. Measurements are presented that compare the directivity of 1-, 2-, and 4-kHz difference tones to that of 1-, 2-, and 4-kHz IM3 lower tones. Furthermore, a cascaded second-order approach for N-element transducer arrays is used to model third-order scattering with good agreement between measurement and theory.

A 65-nm low-noise low-cost ΣΔ modulator for audio applications

This paper introduces a low-noise low-cost ΣΔ modulator for digital audio analog-to-digital conversion. By adopting a low-noise large-output swing operation amplifier, not only is the flicker noise greatly inhibited, but also the power consumption is reduced. Also the area cost is relatively small. The modulator was implemented in a SMIC standard 65-nm CMOS process. Measurement results show it can achieve 96 dB peak signal-to-noise plus distortion ratio (SNDR) and 105 dB dynamic range (DR) over the 22.05-kHz audio band and occupies 0.16 mm2. The power consumption of the proposed modulator is 4.9 mW from a 2.5 V power supply, which is suitable for high-performance, low-cost audio codec applications.